asterisk users - Jun 2004 - Why? oh why can't I dial out?

I have been struggling with my Asterisk setup for 3 days now and I think I 
have done well...apart from the small detail that I cannot dial out on my 
phone (PSTN) line.

My setup is:

Suse Linux 9.0
1 fxo card connected to a BT(UK) line
1 Cisco ATA186 sip v3.0 with two analogue phones attached to it
Asterix CVS-HEAD-05/30/04-06:56:31
with the UK Userid patch applied. Asterisk loads without any warnings.

The problem?
I can receive calls with the userid reported correctly. I can forward them 
to the two SIP ATA lines. I can dial internally (between the two phones) BUT
I cannot dial out :-(

I have tried everything (and yes I searched the world using google but 
nothing seems to apply to my case). So can somebody please direct me to 
possible causes.

The scenario is: if I dial 9123 (for the UK clock) then  output from the 
console is:

   -- Executing Dial("SIP/5000-96f1", "Zap/1/123") in new
stack
     -- Called 1/123
     -- Zap/1-1 answered SIP/5000-96f1
     -- Hungup 'Zap/1-1'

SIP/5000 is one of my ATA phones
ZAP/1 is the fxo card

The call is transferred to Zap/1 as I can hear the dial tone but then 
nothing happens (it does not dial 123). It just stays on the tone until it 
times out. I also tried pressing buttons on my ATA phone but nothing is 
transferred. HELP!




Here is a collection of my conf files:

zaptel.conf

fxsks=1
loadzone=uk
defaultzone=uk

---------------------------
zapata.conf


[channels]
;
; Default language
;
language=en
;
; Default context
;
;context=default
context=incoming
switchtype=national
signalling=fxs_ks
usedistinctiveringdetection=no
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
ancallforward=no
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=0.0
txgain=4.0
immediate=no
musiconhold=default
busydetect=no
callprogress=no
usecallerid=uk

channel => 1

--------------------------------

part of extensions.conf

[incoming]

exten => s,1,SetCallerId(${CALLERID})
exten => s,2,dial(SIP/5000&SIP/5001,10,tr)
exten => s,3,Voicemail,u1000
exten => s,102,Voicemail,b1000


exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1})
exten => _9.,2,Congestion




Vassilis

On Sun, 27 Jun 2004, Vassilis Konstantinou wrote:
> I have been struggling with my Asterisk setup for 3 days now and I think I
> have done well...apart from the small detail that I cannot dial out on my
> phone (PSTN) line.
> [snip]
> The scenario is: if I dial 9123 (for the UK clock) then  output from the
> console is:
>
>    -- Executing Dial("SIP/5000-96f1", "Zap/1/123") in
new stack
>      -- Called 1/123
>      -- Zap/1-1 answered SIP/5000-96f1
>      -- Hungup 'Zap/1-1'
>[snip]
>
> [incoming]
>
> exten => s,1,SetCallerId(${CALLERID})
> exten => s,2,dial(SIP/5000&SIP/5001,10,tr)
> exten => s,3,Voicemail,u1000
> exten => s,102,Voicemail,b1000
>
>
> exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1})
> exten => _9.,2,Congestion
So, assuming that calls from your SIP device are in the same context as
the above extensions, all extensions beginning with a 9 should be dialled
on ${CONSOLE}. On my box, ${CONSOLE}=console/dsp... the sound card. Is
yours set to something similar (or is it really set to dial the zap
interface?)

Not being from the UK myself, I don't know whether the clock's number is
123 or 9123. If it's 9123, then you should be dialing 99123 in order to
get through your dialplan with the 9123 still intact to send to the PSTN.


Greg